| Commit message (Collapse) | Author | Age | Files | Lines |
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The xcb_atom_get functions were removed from xcb-util. Changed these to
xcb_intern_atom/xcb_intern_atom_reply. Also, STRING is now
XCB_ATOM_STRING.
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With automaticl filter loading by module-filter-apply, setting the
virtual sink/source to have the "phone" intended role will break routing
when you first connect a phone stream to an ALSA device and then turn on
your Bluetooth headset. This happens because module-intended-roles
doesn't move a stream if it is already on a device that provides the
required role.
This patch introduces a "manual_load" parameter that is meant to be used
when not using module-filter-apply for loading the AEC module. If this
parameter is set, the virtual devices are given the "phone" role, else
we count on module-filter-heuristics to do the right thing.
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This makes the core code in the filter-* modules generic enough to be
used on sources or sinks. We need special handling for modules that
introduce more than one sink (for now echo-cancel only).
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This makes sure that we handle streams moving between sinks properly. To
do this, we change the way the filter.* properties are handled a little
bit.
Firstly, this splits up the "filter.apply" property into two properties
- "filter.want" and "filter.apply". "filter.apply" acts as before - it
bypasses module-filter-heuristics and directly tells module-filter-apply
what filters are to be applied.
"filter.want" is used to tell module-filter-heuristics what filters the
client wants. The module then decides whether to actually apply the
filter or not (for now, this makes sure we don't apply echo-cancellation
even if requested on phone sinks (where it is assumed AEC is taken care
of or is not required).
Next, we also make sure that we track whether the client set
"filter.apply" or module-filter-heuristics did - and in the latter case,
we recalculate "filter.apply" and then have module-filter-apply apply
the filter if required. This introduces some evil in the form of causing
the move_finish callback to possibly trigger another move, but we
protect for this case (with a property) to be doubly sure of not causing
an infinite loop.
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This makes sure that we don't apply AEC on sinks that are already
connected to a "phone" device, the assumptiong being that anything
marked as such either doesn't have need it, or handles it itself.
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When running two connect-stress tests at the same time the liklihood of >32 streams
per sink increases. All it takes is for an event sound to fire to trigger an abort of
the test.
This leaves just a little bit of wriggle room for a couple external streams.
Of course the overall problem is still there but this just makes it
slightly less likely without really affecting the test itself.
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There were several memory leaks. In addition to those,
pa_dbus_protocol_add_interface() used a string from the
caller as a key to a hashmap, instead of a copy of the
string. This caused trouble when the caller freed the
string while the key was still in use in the hashmap.
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This makes gdb's "info threads" better understandable
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This fixes a long standing race condition when tearing down streams on
Mac OS X.
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Newer generations of libpthread have functions to set and get the thread
names. If available, use them.
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Defining this macro on a global level is disadvantageous for other APIs,
and as we need it for clock_gettime() only on Mac OS X, define it
locally in pulsecore/core-rtclock.c only.
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This value is passed on to the instances of module-coreaudio-device that
are loaded upon device detection. The value is purely optional, as the
device module will fall back to to its default if it's not given.
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This fixes asserts beeing hit when see requests are posted via the
native protocol.
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Adds an autoclean option (defaults to TRUE) that controls whether
module-filter-apply cleans up unused modules or not. This is useful in
cases where you know that a filter will be used often and thus can avoid
overhead from repeated module load/unload.
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1. Remove the "has_destroy_function=false" attribute. It was only
necessary because of a bug in vala which is fixed in 0.12. [1]
2. Add sizes to all fixed-size arrays to make vala recognize them as
such. Using symbolic constants for this is not yet supported. [2]
3. CardInfo struct: Move the brackets in the list of available profiles
to the type to make it clear that this is a dynamically-sized array. [3]
[1] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=622773
[2] https://bugzilla.gnome.org/show_bug.cgi?id=647788
[3] http://0pointer.de/lennart/projects/pulseaudio/doxygen/structpa__card__info.html
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This makes the volume tests run in two loops and print the minimum,
maximum and standard deviation of readings from the inner loop. This
makes it easier to reason out performance drops (i.e. algorithmic
problems vs. other system issues such as processor contention).
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This avoids an assert in pa_sink_input_get_volume() when connecting a
passthrough stream.
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For systems which have a fcntl() implementation, we can simplify the
code which determines whether a file selector is valid in pa_poll().
The old code, which is harder to read and more expensive, stays around
for all platforms we need to emulate poll() for using select(), and
which don't provide fcntl(). IOW, for Windows.
On Mac OS X, however, the detection for bad fds via more select() calls
doesn't work, resulting in hung main loops, so the patch fixes a real
bug there.
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Checking just the flag doesn't work if the sink uses volume sharing, because
such sinks never have PA_SINK_FLAT_VOLUME set.
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filters.
At present the only heuristic is one to apply the echo-cancel filter
when dealing with phone streams.
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streams) based on sink-input property hints.
This module does not yet deal with modules that need matched inputs/outputs
(i.e. echo-cancel) but this will be added in due course.
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This brings more uniformity to arguments to match module-echo-cancel
(which needs both sink and source masters, hence the disambiguation).
This will allow other modules to load filters in a more uniform way
in the future without kludges to deal with variation in arguments.
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When volume changes in bluetooth device PulseAudio volume is rounded
one too low, so if bluetooth headset changes volume and that volume
is immediately set again for bluetooth device, bluetooth step drifts
lower all the time. Volume is incremented by one in the conversion so
that we get right bluetooth step when re-applying volume.
Signed-off-by: Juho Hämäläinen <ext-juho.hamalainen@nokia.com>
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This fix was done for _set_port_cb() already, but the first fix didn't fix
setup_mixer(). Now that's done too.
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used by default.
This module implements a simply policy decision that any newly plugged
in devices should be used.
This is a reasonable approach and paprefs will be updated to allow for
this option to be turned on or off.
This is more or less a stop-gap solution. When priority lists are
implemented in the core, then policy modules may ultimately be
re-engineered to adjust the priority lists rather than doing any of
their own routing per-se.
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This test is based on a threaded main loop and was written to hunt an
evil race condition.
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Previously the userdata for the volume callbacks was saved to
pa_core.shared only once when loading module-bluetooth-device, and only when
the SCO over PCM feature was used. That breaks volume handling in cases where
the HSP profile is used without the SCO over PCM setup. Now the userdata is
set always when a sink or source is created, and removed when a sink or source
is removed.
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The current implementation is totally bogus, it cast the over_sink
userdata to the bluetooth-device userdata... It was failing nicely
because the previous code had a gentle safe-guard in u->profile ==
PROFILE_HSP, and u->profile was just random.
There is no easy way to associate additional data to a sink or
source. Two solutions seems possible: looking up loaded modules and
check which one was handling the sink/source, or using pa_shared. I
went for the second solution.
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Note from Tanu Kaskinen: I resolved some conflicts with newer upstream code, so
if this patch is broken, blame me..
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The #ifdefs only added clutter. I don't see any reason to not compile the
SCO over PCM support in all the time.
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This pulls a2dp-codecs.h from BlueZ which contains the capabilities
structures for SBC and MPEG. We currently have these manually added to
ipc.h, so pulling this header makes our files identical to upstream.
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snd_mixer_selem_get_*_dB_range() and _ask_*_vol_dB().
The check is inspired by a driver that returned higher dB limit from
snd_mixer_selem_get_playback_dB_range() than what _ask_playback_vol_dB()
returned at maximum integer volume.
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e->masks.
SND_MIXER_SCHN_UNKNOWN is defined as -1, so that's not a good array index...
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This is just a quick hack to prevent array overflow. Correct fix would be to
implement support for more channels.
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